1. Field of the Invention
The present invention relates to a communication node and a packet transfer method for transferring a segmented packet, and more particularly to a communication node and a packet transfer method for transferring a packet over a radio network which is a part of global packet-switched network (e.g., Internet).
2. Description of the Background
Recently, demand for radio communications has experienced an explosive increase. An infrastructure for radio communications, represented by cellular phones and PHS (Personal Handy Phone system), has been constructed at a drastically increased pace in recent years. Not only has voice communications been the focus of attention, data communications has attracted significant interest in the form of the Internet.
Future trends of communications networks include (1) a further increase in capacity of communications networks, (2) more widespread use of the so-called multimedia communications in which audio, video, data, etc. are integrated, and (3) an increase in Internet applications as well as more widespread use of the Internet.
Examples of the first trend include next-generation cellular phones (e.g., IMT-2000) and radio ATM networks. Standardization of radio ATM networks has been championed by MMAC. Examples of the second trend include standards (such as H.324) for TV phones. Examples of the third trend include the World Wide Web (WWW), Internet telephony, video-on-demand on the Internet, etc.
It should be noted that the above trends are not independent of one another, but will progress in a closely correlated fashion. For example, TV phones will be able to be implemented as an Internet application, and a radio infrastructure providing Internet services, etc. will emerge.
Key factors in enabling the construction of an infrastructure to support such applications as TV phones on the Internet in a mobile environment, include video/audio coding techniques (e.g., MPEG4), and real-time Internet protocols (e.g., RTP (Real-Time Transport Protocol)). MPEG4 provides coding of video and audio information in a network environment where bandwidth constraints are a major concern, such as phone lines and radio lines, by utilizing a highly efficient coding technique. From a protocol perspective, RTP is useful in operating video and audio applications in a network infrastructure where packets are susceptible to omission and delay, such as the Internet. A combination of these techniques (e.g., MPEG4 over RTP) is expected to realize in the bandwidth constrained Internet multimedia communication.
However, the following problem arises in realizing such Internet multimedia communication. When transmitting video and audio in accordance with MPEG4 in the radio environment, the efficiency of the coding scheme and the error resistance provide the capability to decode the video/audio with acceptable quality to the users, even if some data is omitted or arrives with bit errors during the transmission. In the case of where MPEG4 video/audio data is transmitted as Internet packets in a radio environment, the MPEG4 portion is resistant to errors, but the header portion (that is referred to and used by the network), such as the IP header and the UDP header, has no error resistance capability. If a bit error occurs in the header portion, the Internet packet including such a bit error must be discarded.
As described above, when a packet with a payload that includes data having certain burst error resistance and bit error resistance (e.g., MPEG4 video/audio) is transmitted in the radio environment, the entire packet must be discarded if a bit error occurs in the header portion of the packet. This is problematic because the packet is unnecessarily discarded if the payload can recover lost data, resulting in reduced system throughput.